[Asterisk-Users] Newbie SIP question

; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to localnet = 192.168.1.0 ; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask context = default ; Default for incoming calls [malfoy] type=friend username=malfoy fromuser=Ed Greenbe

[Asterisk-Users] Newbie SIP question

; Default for incoming calls [malfoy] type=friend username=malfoy fromuser=Ed Greenberg secret=2368 host=dynamic defaultip=192.168.1.99 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNS

Re: [asterisk-users] dial a pager and enter DTMF

Supa wrote: Probably just a simple syntax issue, but does anyone know how to dial a number and the once phone has been answered, play DTMF tones and then disconnect. I am trying to use this for page notification. Ive been trying the following string with out luck: exten => s,2,Dial(SIP/TelaSi

Re: [asterisk-users] Sending Email From the dialplan

Al Bochter wrote: I have looked around with no luck. Does anyone know of a way to send an email from the dialplan. The system that I am working on has none thing to do with VoiceMail. This is something like the SMS command but using sending email I am working on a prepaid alarm dispatch progra

Re: [asterisk-users] SIP RTP Tunnel

Also set canreinvite=no between Asterisk and the provider. [EMAIL PROTECTED] wrote: Hola Sanjay, this works pretty well in one direction. The Sip User who is registered at the Asterisk. But the Sip user who calls from sipXYZ.com still sends it data diretly to sip user 1. Any idea? Thanx!!

Re: [Asterisk-Users] Connecting Asterisk to a SIP Gateway

First, you define the destination gateway in sip.conf. Then you write a dialplan in extensions.conf that executes a dial command sending the calls there. Best to read about sip.conf and extensions.conf in www.voip-info.org. --On Friday, April 08, 2005 2:49 PM -0400 "Karras, Darin" <[EMAIL PROTE

[Asterisk-Users] Accountcode in SIP.CONF not set

Using asterisk 1.0.5. In my sip.conf I have this: [user1] type=friend host=dynamic context=users disallow=all allow=ulaw secret=xxx username=user1 accountcode=test In my extensions.conf, I have this: [users] exten => _1NXXNXX,1,NoOp(${ACCOUNTCODE}) exten => _1NXXNXX,2,Dial(SIP/[EMAIL PROTEC

RE: [Asterisk-Users] Sipura 3000 FXO with Asterisk

Hi Razza, I don't know what country you are in, or what your country's telephone numbers look like, but it seems from your dialplan that if you dial an outside number it needs to start with 0X. So if you dial 012345, the Sipura will dial 012345 on the fxo port. If your line needs to dial 12345,

Re: [Asterisk-Users] VOIP to PTSN provider

800 numbers are free to the caller because the recipient pays the charge. Voipjet has no way to get paid anything for carrying the calls, hence they are unwilling to use their resources to move calls with no revenue. Can you blame them? :) --On Saturday, April 16, 2005 9:44 PM +0100 Chris Hill

[Asterisk-Users] Turn off Music on Hold

I'm getting these: Apr 26 12:59:02 NOTICE[14775]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! Apr 26 12:59:02 WARNING[14775]: res_musiconhold.c:205 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3' Apr 26 12:59:02 WARNING[14775]: res_musiconhold.c:278 monmp3threa

Re: [Asterisk-Users] Warm standby boxes - keeping config syncronised?

How about using rsync on a cron timer to copy the config directory from the primary to the secondary. You could have them sync every five minutes if you liked. Also, plug the nic into the secondary (with the same IP address) but just don't autostart the interface. If the first box goes down, y

[Asterisk-Users] noload res_musiconhold.so breaksa IAX

In response to a previous question about disabling music on hold, I was advised to do: noload => res_musiconhold.so Unfortunately, this keeps Asterisk (1.0.5) from running: [chan_iax2.so]Apr 27 04:11:34 WARNING[19654]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_iax2.so: undef

Re: [Asterisk-Users] noload res_musiconhold.so breaksa IAX

Works, thanks. --On Wednesday, April 27, 2005 6:36 AM -0500 Eric Wieling aka ManxPower <[EMAIL PROTECTED]> wrote: Ed Greenberg wrote: In response to a previous question about disabling music on hold, I was advised to do: noload => res_musiconhold.so Unfortunately, this keeps Asteri

[Asterisk-Users] Asterisk on Macintosh - no sound card support?

I installed Asterisk on my Mac and discovered that it had no sound card support out of the box. Am I missing something? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update o

Re: [Asterisk-Users] Polycom IP500

--On Monday, December 06, 2004 7:11 AM -0600 Rich Adamson <[EMAIL PROTECTED]> wrote: The only way to get firmware from Polycom (their policy, not mine) is to become a certified reseller. Check the asterisk archives as someone has posted a site several times with current firmware. Its probably i

Re: [Asterisk-Users] dialplan

--On Monday, December 06, 2004 2:48 PM -0800 Melbourne Lewis <[EMAIL PROTECTED]> wrote: i would like to setup asterisk for any call that comes in on a x100p is answered and is automatically connected to a sip destination external to my asterisk. what is the best dialplan to use? thanks in advan

[Asterisk-Users] Return code from queue app

The Queue app says that queue will return a value (zero or -1). How do I test for this value? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.

[Asterisk-Users] phpconfig - can't locate any of my sections

I downloaded phpconfig and set it up to read my config files, but it never successfully recognizes any of my sections. The regular expression seems to be included in the line: if(preg_match("/^\s*\[([^\]]*)\].*[\r\n]\$/", $line)) and later, the same regex. I'm not sure about the [\r\n] on the en

[Asterisk-Users] Calculating required bandwidth

I was posed this question: A T1 set up for voice carries 24 conversations on a circuit that is 1.544 megabits/second. Right? Well, if you set that T1 up to carry data and run a link between two IP networks over it, how many SIP conversations could it be expected to carry? How about IAX? How wo

RE: [Asterisk-Users] Calculating required bandwidth

--On Thursday, December 16, 2004 3:59 PM -0500 Jim Van Meggelen <[EMAIL PROTECTED]> wrote: I've always found Newton's Telecom Dictionary to be a great reference. It's not too technical, packed with humour, and very comprehensive. I have a very old copy of this, so went off to Amazon to see about

Re: [Asterisk-Users] Calculating required bandwidth

--On Thursday, December 16, 2004 9:45 AM -0800 Ed Greenberg <[EMAIL PROTECTED]> wrote: I was posed this question: I've learned a ton, in the discussion that followed this question. Thanks, all. ___ Asterisk-Users mailing list [EMAIL PROT

Re: [Asterisk-Users] Intercept and redirect outgoing calls ?

--On Friday, December 10, 2004 8:49 PM +0100 Robert Rozman <[EMAIL PROTECTED]> wrote: - Original Message - From: "Nick Barnes" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[EMAIL PROTECTED]> Sent: Friday, December 10, 2004 5:11 PM Subject: RE: [Ast

[Asterisk-Users] 636 Area Code Asterisk Compatible DIDs

Anybody know of good reliable Asterisk compatible DIDs in the 636 (Missouri, USA) area code? Voicepulse doesn't go there, and Broadvoice seems unreliable in my Asterisk installation -- so I'm reluctant to recommend it. Thanks, ___ Asterisk-Users mail

Re: [Asterisk-Users] Asterisk up & running, now what?

Cool things to do for home/small business use... 1. Bring up voicemail on your extensions. 2. Get a US phone number free from ipkall.com. Get some more free numbers from various places. Maybe even pay for one or two. Call forward your POTS lines on busy, so that people can be sent to voicemail o

Re: [Asterisk-Users] Voipjet problems

I made a few voipjet calls today and they all went through just fine. --On Wednesday, December 15, 2004 9:26 PM -0300 Gustavo Russo <[EMAIL PROTECTED]> wrote: Anybody is experimenting problems with Voipjet lately ? Last 2 days we are having some intermitent problems in which, after accepting the

Re: [Asterisk-Users] Bug, Feature, or Limitation?

--On Tuesday, December 21, 2004 9:37 AM -0700 Steve Murphy <[EMAIL PROTECTED]> wrote: Howdy-- I'm playing with different IAX softphones. I've got DIAX and IAXPHONE on a windows (XP) machine on my network, and I'm running asterisk on a newly loaded Fedora Core 3 machine. I set up a separate IAX a

Re: [Asterisk-Users] SPA-2000

Which codecs do you use for the second call? One limitation of the sipura 2000 is that you can not use both ports at the same time with the G729 codec, I belive this may be due to the sipura having an smal CPU that can not handle the load of 2 G729 codecs. Other limitations are the lack of GSM, an

[Asterisk-Users] Play tone till first digit read

I'd like to collect digits (either a fixed number of digits or a variable number ended by a pound sign. Read() is perfect. Authenticate is pretty good too. What I need is to prompt, not with a recording, but by playing a tone that will terminate at the receipt of the first digit, sort of like

[Asterisk-Users] Good 800 Number provider

--On Thursday, February 03, 2005 2:20 PM -0500 Andrew Thompson <[EMAIL PROTECTED]> wrote: What you are seeing with these bargain providers is they have a clause in their contract that says they own the number, not you. It is a lock, and it ought to be illegal, but sadly, it's probably not. If yo

Re: [Asterisk-Users] Wireless LANs and Asterisk

I am doing this with an ancient WAP-11 in Access Point Client Mode. I have it connected to a Sipura 2000 via a crossover cable. It's been very reliable and clean to talk on. I have a WET-11 on order. If it doesn't do a quality job, back it goes. --On Thursday, February 10, 2005 4:17 PM -0800 Sc

Re: [Asterisk-Users] Asterisk Versioning

--On Wednesday, February 09, 2005 5:50 PM -0500 Leif Madsen <[EMAIL PROTECTED]> wrote: On Wed, 9 Feb 2005 20:54:51 +0200, Walid Azab <[EMAIL PROTECTED]> wrote: Just want to understand the difference between Asterisk Versions and please correct me if I am wrong, I understand they are: Stable CVS

Re: [Asterisk-Users] SIP in the Philippines

Can they ping the box successfully? Do they have enough bandwidth? Are you seeing ANY failed or successful registrations? You can change the SIP port in your SIP.CONF, though I don't know if you can use both ports at the same time. Perhaps. Worth reading the wiki to see. --On Friday, February 1

Re: [Asterisk-Users] Q: Does anyone have a WE multi-line card dialer phone working with *?

I have a similar card dialer - a touchtone five line one - in my closet. I checked and there is a good market on eBay for these. Several hundred dollars, last time I looked. I haven't gotten around to advertising mine yet. --On Sunday, February 13, 2005 6:46 PM -0700 George Cohn <[EMAIL PROTEC

Re: [Asterisk-Users] Queue strategy

This begs a question that a fellow over on Asterisk-Biz asked me. If an agent doesn't accept a call, or doesn't answer in an agent-callback scheme, is there a good way to log him out (and email his boss :) Perhaps an AGI script, if one could capture the event. Anybody ever address this? --On Tue

[Asterisk-Users] Difference between a TE410P and TE405P?

Can anybody tell me the difference between a TE410P and a TE405P? Is it JUST the 5v vs 3.3v pcis slot spec, or is there some thing else? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-

Re: [Asterisk-Users] Hi Newbie question

Yes, but he can buy overseas VOIP temination a heck of a lot cheaper than just calling overseas from Nebraska. He may also be able to start overseas DIDs that route to his box here in the states. Rakesh, if this is what you have in mind, let us know and we'll point you in the right direction.

[Asterisk-Users] sending traffic to LiveVoip

I have several DIDs (working well) with LiveVoip and I just signed up for some outbound minutes. Unfortunately they did not send connection instructions. I tried: exten => _1NXXNXX,2,Dial(IAX2/userid:[EMAIL PROTECTED]/${EXTEN}|60|s) but I get the error Feb 19 15:14:09 WARNING[21453]: chan_

Re: [Asterisk-Users] Hi Newbie question

How does he get his offshore relatives into FWD? Nobody said that they have broadband. Just telephones. --On Saturday, February 19, 2005 3:15 PM -0600 Rich Adamson <[EMAIL PROTECTED]> wrote: Or, he could just sign both ends up with FWD and not have to mess with this *.

Re: [Asterisk-Users] sending traffic to LiveVoip

setup. I had the same problem where my DID was setup before my outgoing account. On Sat, 19 Feb 2005 13:16:15 -0800, Ed Greenberg <[EMAIL PROTECTED]> wrote: I have several DIDs (working well) with LiveVoip and I just signed up for some outbound minutes. Unfortunately they did not sen

Re: [Asterisk-Users] wiki down?

--On Sunday, February 20, 2005 10:38 PM + Peter Bowyer <[EMAIL PROTECTED]> wrote: On Sun, 20 Feb 2005 22:45:42 +0100, Roy Sigurd Karlsbakk <[EMAIL PROTECTED]> wrote: It seems to me wiki downtime is somehow regular. Is this the fact? If so, should it be moved? Just to add some balance to this

[Asterisk-Users] LiveVoip digit loss

Receiving calls from LiveVoip DIDs results in dropped DTMF digits. I'm using SIP, not IAX, and I've tried this without a dtmfmode and with dtmfmode in all the various permutations. Note that LiveVoip does not instruct us to put any dtmfmod statement in. The server is set to do ulaw and I've veri

Re: [Asterisk-Users] Zaptel Red Alarm

There are many programs that will tail one or more log file looking for specific patterns and doing notifications and actions on them. I don't have one handy, but they are out there, both free and commercial. Perhaps somebody else will chime in... --On Wednesday, February 23, 2005 8:33 AM -0600

Re: [Asterisk-Users] call pickup with Sipura-3000

When I pick up calls on my Sipura I just dial *8# instead of *8. The # will end the Sipura's dial plan. If you put *8 into the dialplan, that would work too. --On Saturday, February 26, 2005 11:39 PM -0700 Joseph <[EMAIL PROTECTED]> wrote: On Sat, 2005-02-26 at 18:22 -0700, Joseph wrote: I can no

Re: [Asterisk-Users] VoipSupply.com

--On Tuesday, May 17, 2005 5:24 PM -0700 Manjit Riat <[EMAIL PROTECTED]> wrote: I am going to buy some IP phones from them but I sent them an email couple of weeks ago and got no reply. Has anyone ordered anything from them? Any other places that I can buy from? Sorry if it's a wrong post. I ha

RE: [Asterisk-Users] need 7960 power cubes

I will chime in and say that I am running with VoipSupply's OEM cube. The Cisco phone is very happy. --On Wednesday, May 18, 2005 6:04 PM -0400 Garrett Smith <[EMAIL PROTECTED]> wrote: Rafal: We have OEM power cubes for Cisco phones for $16.95 each. Thanks, Garrett Smith <[EMAIL PROTECTED]> 716

[Asterisk-Users] (another) cisco 7960 question

My 7960 is configured for two lines, and I can turn the other appearance buttons into speed dials from the menus, but is there any way to program the speed dials in the SIP.conf file? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com ht

Re: [Asterisk-Users] Random Sound File

AGI script in Perl (or whatever) should do it easily. Get a random number between 1 and n then index into a list of files. Then either play the file from the AGI script or just return it's name in a variable for later playing. --On Tuesday, May 24, 2005 2:01 PM +0100 "Marshall, Ed"

Re: [Asterisk-Users] origination providers

Hi Mike, Understand that your supplier will be paying by the minute. What you want is your suppliers worst nightmare. Fixed income and variable (increasing) costs, both to his upstream provider and also in bandwidth, both network and computer. Many of us will be happy to supply you with as m

Re: [Asterisk-Users] Asterisk Box as a Router, Firewall and DHCP Server

El mié, 01-06-2005 a las 22:27, Samy Antoun escribió: I'm planning to get my Asterisk box out of the LAN, get rid of my router and make the box acts as a Router, Firewall, DHCP Server (with Shorewall). I'll do that to be able to use some SIP clients remotely. Does anyone doing the same wit

Re: [Asterisk-Users] Extension 'hint' info please?

--On Saturday, June 04, 2005 10:41 PM -0700 Chris Coulthurst <[EMAIL PROTECTED]> wrote: I have a Polycom 500 and would love one of the line-appearance keys to show me if a certain person/people are on the phone upstairs. This 'hint' priority seems to have little-to-no documentation. So, if

RE: [Asterisk-Users] Asterisk code

This advice is good. My first change to the asterisk code was to copy the DISA application and change it. So I have my own DISA application (DISA10) that takes ten digits instead of 'n' digits + pound. From this I learned how to add the new application to Asterisk, how to recompile and reinst

Re: [Asterisk-Users] What my IAXy could have been...

Sipura 1000 or 2000? --On Tuesday, March 01, 2005 10:15 PM +0900 Daiku <[EMAIL PROTECTED]> wrote: Hi, methinks that in the good 3 months since i ordered an IAXy, things have changed so much that now almost anybody out there with a VoIP hardweare website offers complete phones for less money than

Re: [Asterisk-Users] Re: No ringback over IAX - LiveVoip

--On Friday, March 04, 2005 11:58 AM -0600 James Taylor <[EMAIL PROTECTED]> wrote: It would be nice if they told us what the "problem" with Asterisk is... There's probably enought great minds on this list, that it could be resolved. There is clearly an issue between LiveVoip and Asterisk. The Li

[Asterisk-Users] Agents and Queues

I have a queue and some agents. The agents are not logged in but are members of the queue. Now, joinempty=no I would assume that you cannot join this queue but will fall into the goto(queuefail|1|1). This is not so. I join the queue just fine. If I remove the three members from the queue (comm

Re: [Asterisk-Users] SJphone on PDA registering with Asterisk???

Debugging lack of registration: Watch the console (set verbose 255) and see if there are registration attempts. If you see failures, the name and secret are probably wrong. If you don't see attempts, either the phone isn't trying, or there is a connectivity problem from the phone to the Asterisk

RE: [Asterisk-Users] IAX2 800 Termination

I have a LiveVoip DID which is in daily use feeding an Asterisk box as a conference bridge. It has been flawless. I may have had issues with LiveVoip on my toll DIDs, but the 800 service has been fine. --On Thursday, March 10, 2005 4:04 PM -0700 Wiley Siler <[EMAIL PROTECTED]> wrote: I am b

[Asterisk-Users] Agents without agent channel

Has anybody used the dialplan, "Agents without agent channel" found at Did it work for you? Did it need much customization. I already note that it requires a res_perl routine (or the removal of the call to same.

Re: [Asterisk-Users] Queue drop out into context not working?

I just made this work without much difficulty, so I imagine you can do so too. It does not look like anything wrong with your queue entry. Can you post your [testqueue-drop] context from extensions.conf? --On Tuesday, March 15, 2005 9:02 AM +0100 "Sascha E. Pollok" <[EMAIL PROTECTED]> wrote: G

Re: [Asterisk-Users] Asterisk Queue strange behaviour

Try adding an agent and logging into that agent (rather than making the SIP phone the member, directly). --On Tuesday, March 15, 2005 1:19 PM +0100 Jan Marius Evang <[EMAIL PROTECTED]> wrote: Hi. I have a problem which I assume would be easy to fix, but I can't find anything about it... I wish

Re: [Asterisk-Users] Three way calling with X-Lite / MeetMe

Try SJPhone. (www.sjlabs.com) --On Tuesday, March 15, 2005 3:00 PM + Chris Blunt <[EMAIL PROTECTED]> wrote: Hi All, Does any one know of a way to make a three way call from Asterisk using X-Lite. I need the ability to be able to call someone on the PSTN using my IAX provider then bring ano

[Asterisk-Users] Database families and keys

When I do a database show, I get something like this: /SIP/Registry/202 : 192.168.1.89:5060:3600:202:sip:[EMAIL PROTECTED]:5060 /SIP/Registry/203 : 192.168.1.89:5060:3600:203:sip:[EMAIL PROTECTED]:5060 /SIP/Registry/211 : 24.184.18.140:5060:60:211:sip:[EMAIL PROTECTED]:5060 /SIP/Registry/212 :

Re: [Asterisk-Users] PSTN > Voicemail

It's not a stupid question, but answering it requires some more info about your setup. If you have any kind of auto-answering DID which allows you to then tone in an extension, you can add an extension to that (any number that fits the dialplan) which connects you to voicemailmain. If you don'

Re: [Asterisk-Users] * and DirecWay

--On Saturday, March 19, 2005 8:45 AM -0800 Bruce Komito <[EMAIL PROTECTED]> wrote: If you have any experience using * (or VoIP in general) with DirecWay, please respond privately. I am particularly interested in experiences in Latin America. TIA! I have an Asterisk system on the net, and I tri

Re: [Asterisk-Users] Cisco 7960 SIP Firmware

I wonder if VoipSupply can sell the maintenance contract for the phone? Wouldn't hurt to ask. The fellow from VS is a regular poster over on the asterisk-biz list. --On Saturday, March 19, 2005 11:09 AM -0600 Jerry <[EMAIL PROTECTED]> wrote: I would suggest contacting a dealer until you find o

Re: [Asterisk-Users] DISA -> macro = congestion

I see a missing left parentheses on the line that starts exten => s,2,Dial,IAX2... I do not know if that is causing your problem. Whatever it is, a good troubleshooting technique is to watch the console while doing the operation, with verbose set to 255. --On Saturday, March 19, 2005 12:11 PM

Re: [Asterisk-Users] VoIP service through Asterisk?

--On Sunday, March 20, 2005 1:41 PM +0400 Jean-Michel Hiver <[EMAIL PROTECTED]> wrote: re is: - iax.cc (haven't tried them) - connect.voicepusle.com (haven't tried them) connect.voicepulse.com: Very good for incoming. Too expensive for outgoing. - nufone.net (they're meant to be quite reliable -

[Asterisk-Users] D() option on Dial

Using the D() option on a Dial, I get only the first digit of my string. I'm running 1.0.5. I looked at the change log and didn't see anything that referenced this. Anybody know what I'm doing wrong? exten => s,3,Dial(IAX2/[EMAIL PROTECTED]/13115552368,60,D(1234567890123456789)) Thanks in advanc

[Asterisk-Users] Sipura 3000 FXO with Asterisk

Anybody using a Sipura 3000 for FXO with Asterisk? Mine is working except for one small nit... When a call comes in from the PSTN, the Sipura answers it and then passes it on to Asterisk, which plays extension ring tone. I'd prefer for the POTS line to stay on-hook while the extension rings, and

Re: [Asterisk-Users] Sending DTMF back in a dialed/answered channel before bridging a call

Try the D() modifier to the dial command. --On Friday, April 01, 2005 11:27 AM -0500 BJ Weschke <[EMAIL PROTECTED]> wrote: Is there a uniform way that exists already to send DTMF down a channel that's just been dialed and has answered before bridging the two channels together? (eg. an IVR answer

[Asterisk-Users] SIP/SDP packaged in Multipart/Mixed mime type

Does Asterisk's sip implementation support SDP packaged in Multipart/Mixed mime type? It looks like it does not. Any light to shed out there? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asteri

[Asterisk-Users] Analog ports via USB

inter to an article? Other info? Thanks, Ed Greenberg San Jose, CA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/lis

[Asterisk-Users] Siemens optiPoint 300

are surplus and have no supplies. Thanks, Ed Greenberg San Jose, CA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman

[Asterisk-Users] Cisco 7940 Volume low

and also Zap. Transferring the call to a Zap extension results in a much improved incoming volume. I'm not sure if outgoing volume is behaving the same way. Any suggestions? Any more info I can post to make this more answerable? :) Thanks, Ed Greenberg San Jos

Re: [Asterisk-Users] need some advice

connected to an FXS. (Living room and bedroom - extension 208) A desk phone connected to an FXS (ext 201) A Cisco 7940 two line IP phone on 202 and 203 (borrowed -- too expensive to buy) Several soft phones on PCs, just for experimenting. (205,206) Feel free to question on this. Ed Greenber

Re: [Asterisk-Users] need some advice

Hi Pat, Please see below --On Wednesday, November 24, 2004 6:08 PM +0100 p80 <[EMAIL PROTECTED]> wrote: On Wednesday 24 November 2004 03:39 am, Ed Greenberg wrote: Incoming service via VOIP. Many VOIP services can terminate directly in your asterisk box. Some require that you use

Re: [Asterisk-Users] asterisk and verizon DSL

I have it working with SBCs DSL (San Jose). --On Wednesday, November 24, 2004 7:18 PM -0500 [EMAIL PROTECTED] wrote: Is anyone succesfully running Asterisk behind verizon residential DSL? I seem to be having some problems with my Asterisk server switching to Verizon. I'm attempting to do some trou

Re: [Asterisk-Users] FWD with iax2

You are not the only one. I have the same problem and there's some discussion of it on the freeworld.com discussion forum. --On Saturday, November 27, 2004 10:28 AM +0800 Ronald Wiplinger <[EMAIL PROTECTED]> wrote: iax2 show registry: Host UsernamePercived Refresh

[Asterisk-Users] cisco 7940/60 dialplan

--On Friday, November 26, 2004 11:24 PM -0600 [EMAIL PROTECTED] wrote: Also, as a side query, does anyone know how to set a 7960 SIP to dial without having to hit the Dial key on the phone? I type in the numbers and it sits until I press DIAL, which seems kind of counter-intuitive. This is drive

Re: [Asterisk-Users] asterix as proxy

--On Saturday, November 27, 2004 10:05 AM -0500 Philippe Daoust <[EMAIL PROTECTED]> wrote: Grant Williamson wrote: Hi, I would like to setup an asterix server as a proxy allowing me to, access multiple SIP accounts from a single hardware device. Are there any pointers on setting this up? This is

Re: [Asterisk-Users] very newbie question

Exactly. Create a context that just allows what you want him to do. If this gets complex - with multiple permission levels - then you create smaller context fragments and include them. Best, --On Saturday, November 27, 2004 7:37 PM + Corvin <[EMAIL PROTECTED]> wrote: Hi everyone! I have ve

[Asterisk-Users] Pick up call without ringing an extension

I have a line in my house that goes to phones across the line BEFORE Asterisk. My wife uses them :) I plugged the line into an FXO port and pointed it at an extension, so I can pick it up as well. I can also pick it up with *8. One side effect is that once my wife picks up a call, the Asterisk

Re: [Asterisk-Users] After setting up my FXO card, what should I now order from my telco?

--On Wednesday, December 01, 2004 12:00 AM -0600 Brent Clements <[EMAIL PROTECTED]> wrote: Ok, so I'm setting up my small office. I have my asterisk machine setup and I have 3 sip phones connected as my stations and a 4 port FXO card ready to go(planning on only using 2 lines currently). What sho

Re: [Asterisk-Users] Asterisk for home office

--On Wednesday, December 01, 2004 12:59 AM -0800 Lee <[EMAIL PROTECTED]> wrote: Guess I need to research DID providers a bit more, as I would want a local number, or to keep my existing number...But I'm in a medium sized city, and won't be surprised if local numbers aren't available. Thanks --

Re: [Asterisk-Users] cisco 7940 help

On should watch the system log on the tftp server while doing this. It may require starting tftpd with a -v or similar. The error messages will be very instructive. --On Wednesday, December 01, 2004 4:12 PM + "K. C. Li" <[EMAIL PROTECTED]> wrote: On Wed, 1 Dec 2004, Andrei (MPI) wrote: Ric

Re: [Asterisk-Users] Fedora Core 2 firewall rules - NO NAT!

This is a common configuration. My asterisk box is also my router. dsl -> eth1 -> linux <- eth0 <- home network The linux box routes and firewalls. In your case, you probably have another router and just want to multi-home your asterisk box. Is this correct? If so, just don't turn on routing, or

Re: [Asterisk-Users] 900# DID?

--On Thursday, December 02, 2004 2:15 AM -0600 Jay Milk <[EMAIL PROTECTED]> wrote: Here's a question I haven't seen asked nor answered on this list: Is there a provider who offers incoming 900# services? I want to establish a 900# to be used in (about 60-70) domain registrations, to deter telem

Re: [Asterisk-Users] Ring all Configured Extension

Can you put the "SIP/3001&SIP/3002&SIP/3003&..." in some sort of variable, macro or other storage? --On Thursday, December 02, 2004 10:05 AM -0600 Matthew Boehm <[EMAIL PROTECTED]> wrote: exten => 4000,1,Dial(SIP/3001&SIP/3002&SIP/3003&..., 30, t) Matthew - Original Message - Fr

Re: [Asterisk-Users] Ouch, part reset, quickly

I have exactly this problem. When it happens, I lose access to some FXS ports and get Geiger counter style clicking on the FXOs. I just opened a ticket with Digium on the subject, but given what I just read, perhaps I should not have high hopes. --On Friday, December 03, 2004 11:33 PM +0800 Ga

[Asterisk-Users] Round Robin Call Distribution

I need to send calls to a bank of servers in a round robin fashion. Has anybody implemented this in dialplan language? If not, in some other fashion. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBS

Re: [Asterisk-Users] Aasterisk large-scale deployment w/analog phones

Consider also a set of AudioCodes MP124s. These are 24 port ATAs. Work well for us. Four of them would give 96 ports. No zap hardware needed to connect the channel banks. --On Wednesday, February 15, 2006 2:14 PM +0200 maka <[EMAIL PROTECTED]> wrote: hello, I am planning a fairly large

Re: [Asterisk-Users] Fwd: Which ATA device do you recommend?

For single and two-port applications, I've had very good luck with Sipura 2000s. Now available as Linksys PAP2-NA. --On Wednesday, February 15, 2006 3:08 PM + Marco Mouta <[EMAIL PROTECTED]> wrote: -- Forwarded message -- From: Marco Mouta <[EMAIL PROTECTED]> Date: Feb

[Asterisk-Users] An array of extensions in my lab

When working on prototypeing asterisk installations, I sometimes need an array of extensions in my lab. How do others handle this? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options

Re: [Asterisk-Users] Fwd: Which ATA device do you recommend?

Michael J. Liberatore wrote: Are the PAP2's you can get branded vonage at staples for free after rebate still hackable? I read that you cant do it beyond a certain firmware but wasn't sure if it had to be connected to the internet for that download or if it ships with that now My und

Re: [Asterisk-Users] asterisk error

I usually see this when doing operations on variables that are blank. In your case, the input is ' + 1'. Clearly there was something to the left of the + but it's blank. If you're adding one to something, make sure there is a number on the left side of the plus sign. Probably by initializing

[Asterisk-Users] A room full of Cisco 7960s behind NAT

I need to set up an office full of Cisco 7960 phones behind NAT with the server out in Colo. The first test phone registers fine, but the second one does not register. The first phone's registration looks like so: /SIP/Registry/3115552368 :64.169.xx.yyy:38836:3600:3115552368:sip:[EMAI

Re: [Asterisk-Users] A room full of Cisco 7960s behind NAT

On Tuesday, February 28, 2006 2:29 PM -0800 Ed Greenberg <[EMAIL PROTECTED]> wrote: I need to set up an office full of Cisco 7960 phones behind NAT with the server out in Colo. The first test phone registers fine, but the second one does not register. The first phone's registratio

RE: [Asterisk-Users] A room full of Cisco 7960s behind NAT

I've tried it both ways with no noticeable change. --On Tuesday, February 28, 2006 7:25 PM -0700 Damon Estep <[EMAIL PROTECTED]> wrote: Your posted config had nat=1, not nat=yes. Are they interchangeable? I thought I remembered nat=1 either doing something a little different or not doing anyt

RE: [Asterisk-Users] A room full of Cisco 7960s behind NAT

--On Wednesday, March 01, 2006 3:00 AM -0600 Rich Adamson <[EMAIL PROTECTED]> wrote: The 7960 SIPDefault.cnf has a line like this: ; nat_enable: 0 ; 0-Disabled (default), 1-Enabled to control the nat function "at the phone". Since I'm not the OP but do have multiple 7960's

RE: [Asterisk-Users] A room full of Cisco 7960s behind NAT

--On Wednesday, March 01, 2006 12:53 AM -0500 Alexander Lopez <[EMAIL PROTECTED]> wrote: I have about 10 cisco 7960/40 behind a Nat router no problem. They work with all features. Alexander... Can you post or mail to me the relevant contents of your SIPXXX.cnf files? Also, can you post

[Asterisk-Users] Speech Recognition

I've been asked to integrate some simple speech recognition with an IVR. Is there anything that people are using with Asterisk for this? Where should I start reading? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.di

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